[asterisk-bugs] [JIRA] (ASTERISK-13145) [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML
Stephen Belton (JIRA)
noreply at issues.asterisk.org
Mon Oct 26 09:03:21 CDT 2020
[ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=252573#comment-252573 ]
Stephen Belton commented on ASTERISK-13145:
-------------------------------------------
I'm trying to get video calling to work with a Cisco DX650. I'm running Asterisk 16.13 with the latest call manager patch. When cisco_usecallmanager=yes the call completes, but I only get audio, not video. When cisco_usecallmanager=no the call completes and I get full audio and video. I think it has something to do with how the patch is handling the rtpmap--when cisco_usecallmanager=yes 3 RTP profiles for H264 are sent: 100, 126 & 97 with the final mapping going to 100, when no only the 97 profile for H264 is sent with final mapping of course going to 97.
{noformat}
***Offer Use Call Manager = Yes***
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=640016;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428016;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428016;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
***Final Mapping***
m=video 17536 RTP/AVP 100
b=TIAS:4000000
a=rtpmap:100 H264/90000
a=fmtp:100 max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000;packetization-mode=1;level-asymmetry-allowed=1;profile-level-id=640016
a=imageattr:100 recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=sendrecv
***Error Logged***
[2020-10-25 20:22:46] WARNING[8547][C-00000001]: chan_sip.c:8455 sip_indicate: Don't know how to indicate condition 36
{noformat}
{noformat}
***Offer Use Call Manager=No***
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428016;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
***Final Mapping***
m=video 19258 RTP/AVP 97
a=rtpmap:97 H264/90000
a=fmtp:97 max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000;packetization-mode=0;level-asymmetry-allowed=1;profile-level-id=428016
a=sendrecv
{noformat}
Is there a way I can force mapping to 97 (assuming that's the problem) or is there a fix that can be made in the next version of the patch? Would love to get video working with this phone they are fairly cheap on eBay and have great video quality.
For reference I'm including the DX-650 configuration file and logs with cisco_usecallmanager set to yes and no.
DX650 Config
{noformat}
<?xml version="1.0" encoding="UTF-8"?>
<device xsi:type="axl:XIPPhone" ctiid="62943" uuid="{e045c922-43ad-2320-24c9-be1f8abc3d0b}">
<fullConfig>true</fullConfig>
<portalDefaultServer></portalDefaultServer>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<ipAddressMode>0</ipAddressMode>
<ipPreferenceModeControl>0</ipPreferenceModeControl>
<redirectEnable>false</redirectEnable>
<echoMultiEnable>false</echoMultiEnable>
<ipMediaAddressFamilyPreference>0</ipMediaAddressFamilyPreference>
<mlppDomainId>000000</mlppDomainId>
<mlppIndicationStatus>Off</mlppIndicationStatus>
<preemption></preemption>
<executiveOverridePreemptable></executiveOverridePreemptable>
<devicePool uuid="{d0181915-1eac-910c-3a0f-f03c26afd832}">
<revertPriority>0</revertPriority>
<name>Phones - 1.5M Video EST EDT</name>
<dateTimeSetting uuid="{daaf53f2-bb03-b274-953c-5090869fc211}">
<name>EST-5</name>
<dateTemplate>M/D/YA</dateTemplate>
<timeZone>Eastern Standard/Daylight Time</timeZone>
<olsonTimeZone>America/New_York</olsonTimeZone>
</dateTimeSetting>
<callManagerGroup>
<name>Default</name>
<tftpDefault>false</tftpDefault>
<members>
<member priority="0">
<callManager>
<name>Asterisk</name>
<description></description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>10.118.1.28</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<srstInfo uuid="{cd241e11-4a58-4d3d-9661-f06c912a18a3}">
<name>Disable</name>
<srstOption>Disable</srstOption>
<userModifiable>false</userModifiable>
<ipAddr1></ipAddr1>
<port1>2000</port1>
<ipAddr2></ipAddr2>
<port2>2000</port2>
<ipAddr3></ipAddr3>
<port3>2000</port3>
<sipIpAddr1></sipIpAddr1>
<sipPort1>5060</sipPort1>
<sipIpAddr2></sipIpAddr2>
<sipPort2>5060</sipPort2>
<sipIpAddr3></sipIpAddr3>
<sipPort3>5060</sipPort3>
<isSecure>false</isSecure>
</srstInfo>
<connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>
<sipProfile>
<sipProxies>
<backupProxy>USECALLMANAGER</backupProxy>
<backupProxyPort>5060</backupProxyPort>
<emergencyProxy>USECALLMANAGER</emergencyProxy>
<emergencyProxyPort>5060</emergencyProxyPort>
<outboundProxy>USECALLMANAGER</outboundProxy>
<outboundProxyPort>5060</outboundProxyPort>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x-cisco-serviceuri-cfwdall</callForwardURI>
<callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
<callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
<meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
<abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
<rfc2543Hold>false</rfc2543Hold>
<callHoldRingback>2</callHoldRingback>
<URIDialingDisplayPreference>1</URIDialingDisplayPreference>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
<retainForwardInformation>false</retainForwardInformation>
</sipCallFeatures>
<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<timerRegisterExpires>60</timerRegisterExpires>
<timerRegisterDelta>0</timerRegisterDelta>
<timerKeepAliveExpires>120</timerKeepAliveExpires>
<timerSubscribeExpires>120</timerSubscribeExpires>
<timerSubscribeDelta>5</timerSubscribeDelta>
<timerT1>500</timerT1>
<timerT2>4000</timerT2>
<maxRedirects>70</maxRedirects>
<remotePartyID>false</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<kpml>3</kpml>
<phoneLabel></phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>true</callStats>
<offhookToFirstDigitTimer>5000</offhookToFirstDigitTimer>
<T302Timer>15000</T302Timer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<poundEndOfDial>false</poundEndOfDial>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<organizationTopLevelDomain>10.118.1.28</organizationTopLevelDomain>
<sipLines>
<line button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel>310</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>3108</name>
<displayName>3108</displayName>
<autoAnswer>
<autoAnswerEnabled>0</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>3108</authName>
<authPassword>secretpassword</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messageWaitingAMWI>0</messageWaitingAMWI>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact></contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>10</maxNumCalls>
<busyTrigger>6</busyTrigger>
</line>
<line button="2" lineIndex="1">
<featureID>21</featureID>
<featureLabel>TestBLF</featureLabel>
<speedDialNumber>3105</speedDialNumber>
</line>
</sipLines>
<externalNumberMask>5555555555</externalNumberMask>
<voipControlPort>5060</voipControlPort>
<ringSettingBusyStationPolicy>1</ringSettingBusyStationPolicy>
<dialTemplate>dialplan2.xml</dialTemplate>
<softKeyFile>softkeys.xml</softKeyFile>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>true</alwaysUsePrimeLineVoiceMail>
</sipProfile>
<MissedCallLoggingOption>10</MissedCallLoggingOption>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>3</callLogBlfEnabled>
</commonProfile>
<loadInformation>sipdx650.10-2-5-215</loadInformation>
<inactiveLoadInformation></inactiveLoadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<allowBTContactImport>1</allowBTContactImport>
<allowBTMobileHandsfree>1</allowBTMobileHandsfree>
<recordingTone>0</recordingTone>
<settingsAccess>1</settingsAccess>
<recordingToneLocalVolume>100</recordingToneLocalVolume>
<recordingToneRemoteVolume>50</recordingToneRemoteVolume>
<recordingToneDuration></recordingToneDuration>
<deviceUIProfile>0</deviceUIProfile>
<detectCMConnectionFailure>0</detectCMConnectionFailure>
<garp>0</garp>
<multiUser>0</multiUser>
</vendorConfig>
<commonConfig>
<ciscoCamera>1</ciscoCamera>
<videoCapability>1</videoCapability>
<rtcp>1</rtcp>
<videoRtcp>1</videoRtcp>
<webProtocol>0</webProtocol>
<requireScreenLock>0</requireScreenLock>
<webAccess>0</webAccess>
<sshAccess>0</sshAccess>
<sendKeyAction>1</sendKeyAction>
<RingLocale>0</RingLocale>
<appInstallFromUnknownSrc>1</appInstallFromUnknownSrc>
<appInstallFromAndroidMarket>true</appInstallFromAndroidMarket>
</commonConfig>
<versionStamp>1387322115-49d5fd49-52b6-4926-b708-11c02cb22c22</versionStamp>
<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en_US</langCode>
<version></version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>Canada</networkLocale>
<networkLocaleInfo>
<name>Canada</name>
<uid>64</uid>
<version></version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<transportLayerProtocol>1</transportLayerProtocol>
<dndCallAlert>5</dndCallAlert>
<phonePersonalization>1</phonePersonalization>
<rollover>0</rollover>
<singleButtonBarge>0</singleButtonBarge>
<joinAcrossLines>0</joinAcrossLines>
<autoCallPickupEnable>false</autoCallPickupEnable>
<blfAudibleAlertSettingOfIdleStation>0</blfAudibleAlertSettingOfIdleStation>
<blfAudibleAlertSettingOfBusyStation>0</blfAudibleAlertSettingOfBusyStation>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
<processNodeName>10.118.1.28</processNodeName>
</capf>
</capfList>
<certHash/>
<encrConfig>false</encrConfig>
<advertiseG722Codec>0</advertiseG722Codec>
<mobility>
<handoffdn/>
<dtmfdn/>
<ivrdn/>
<dtmfHoldCode>*81</dtmfHoldCode>
<dtmfExclusiveHoldCode>*82</dtmfExclusiveHoldCode>
<dtmfResumeCode>*83</dtmfResumeCode>
<dtmfTxfCode>*84</dtmfTxfCode>
<dtmfCnfCode>*85</dtmfCnfCode>
</mobility>
<TLSResumptionTimer>0</TLSResumptionTimer>
<phoneServices useHTTPS="true">
<provisioning>0</provisioning>
<phoneService type="1" category="0">
<name>Missed Calls</name>
<url>Application:Cisco/MissedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Received Calls</name>
<url>Application:Cisco/ReceivedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Placed Calls</name>
<url>Application:Cisco/PlacedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Personal Directory</name>
<url>Application:Cisco/PersonalDirectory</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
</device>
{noformat}
cisco_usecallmanager=yes (audio works, video does not)
{noformat}
<--- SIP read from TCP:10.118.1.165:56009 --->
NOTIFY sip:3108 at 10.118.1.28 SIP/2.0
Via: SIP/2.0/TCP 10.118.1.165:56009;branch=z9hG4bK4495b3ed
To: "3108" <sip:3108 at 10.118.1.28>
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e76013b37f84246-2aeb1b20
Call-ID: 1aa3d235-6b77565b at 10.118.1.165
Session-ID: 007bf9fd00105000a00000cae5415e76;remote=00000000000000000000000000000000
Date: Mon, 26 Oct 2020 00:22:43 GMT
CSeq: 7 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:3108 at 10.118.1.165:56009;transport=tcp>
Authorization: Digest username="3108",realm="asterisk",uri="",response="085446480d6701789aadb99b80515030",nonce="38d18d94",algorithm=MD5
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 353
Content-Type: application/dialog-info+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8" ?>
<dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="0" state="partial" entity="sip:3108 at 10.118.1.165">
<dialog id="6" call-id="00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165" local-tag="00cae5415e76013a71a216aa-2392cbbe"><state>trying</state></dialog>
</dialog-info>
<------------->
--- (17 headers 4 lines) ---
Sending to 10.118.1.165:56009 (NAT)
<--- Transmitting (NAT) to 10.118.1.165:56009 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.118.1.165:56009;branch=z9hG4bK4495b3ed;received=10.118.1.165;rport=56009
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e76013b37f84246-2aeb1b20
To: "3108" <sip:3108 at 10.118.1.28>;tag=as6a180363
Call-ID: 1aa3d235-6b77565b at 10.118.1.165
CSeq: 7 NOTIFY
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1aa3d235-6b77565b at 10.118.1.165' in 32000 ms (Method: NOTIFY)
<--- SIP read from TCP:10.118.1.165:56009 --->
INVITE sip:12345678 at 10.118.1.28;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.118.1.165:56009;branch=z9hG4bK09b6b51a
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e76013a71a216aa-2392cbbe
To: <sip:12345678 at 10.118.1.28>
Call-ID: 00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165
Max-Forwards: 70
Session-ID: 6c22b06500105000a00000cae5415e76;remote=00000000000000000000000000000000
Date: Mon, 26 Oct 2020 00:22:43 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP-DX650/10.2.5
Contact: <sip:3108 at 10.118.1.165:56009;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP00CAE5415E76";video
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Authorization: Digest username="3108",realm="asterisk",uri="sip:12345678 at 10.118.1.28;user=phone",response="c9ae541033b804fdf89c524e093ca275",nonce="38d18d94",algorithm=MD5
Content-Length: 1244
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 14753 0 IN IP4 10.118.1.165
s=SIP Call
b=AS:4064
t=0 0
m=audio 20262 RTP/AVP 0 8 116 18 101
c=IN IP4 10.118.1.165
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 20678 RTP/AVP 100 126 97
c=IN IP4 10.118.1.165
b=TIAS:4000000
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=640016;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428016;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428016;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=content:main
a=label:11
a=sendrecv
<------------->
--- (22 headers 35 lines) ---
Sending to 10.118.1.165:56009 (NAT)
Sending to 10.118.1.165:56009 (NAT)
Using INVITE request as basis request - 00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165
Found peer '3108' for '3108' from 10.118.1.165:56009
<--- Reliably Transmitting (no NAT) to 10.118.1.165:56009 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 10.118.1.165:56009;branch=z9hG4bK09b6b51a;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e76013a71a216aa-2392cbbe
To: <sip:12345678 at 10.118.1.28>;tag=as368526ca
Call-ID: 00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165
CSeq: 101 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer,X-cisco-sis-7.0.0
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="50842d88"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165' in 6400 ms (Method: INVITE)
<--- SIP read from TCP:10.118.1.165:56009 --->
ACK sip:12345678 at 10.118.1.28;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.118.1.165:56009;branch=z9hG4bK09b6b51a
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e76013a71a216aa-2392cbbe
To: <sip:12345678 at 10.118.1.28>;tag=as368526ca
Call-ID: 00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165
Session-ID: 6c22b06500105000a00000cae5415e76;remote=00000000000000000000000000000000
Max-Forwards: 70
Date: Mon, 26 Oct 2020 00:22:43 GMT
CSeq: 101 ACK
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from TCP:10.118.1.165:56009 --->
INVITE sip:12345678 at 10.118.1.28;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.118.1.165:56009;branch=z9hG4bK6087719e
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e76013a71a216aa-2392cbbe
To: <sip:12345678 at 10.118.1.28>
Call-ID: 00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165
Max-Forwards: 70
Session-ID: 6c22b06500105000a00000cae5415e76;remote=00000000000000000000000000000000
Date: Mon, 26 Oct 2020 00:22:43 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP-DX650/10.2.5
Contact: <sip:3108 at 10.118.1.165:56009;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP00CAE5415E76";video
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Authorization: Digest username="3108",realm="asterisk",uri="sip:123456789 at 10.118.1.28;user=phone",response="f408a9c85b5c847ac0e3395a33757d24",nonce="50842d88",algorithm=MD5
Content-Length: 1244
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 14753 0 IN IP4 10.118.1.165
s=SIP Call
b=AS:4064
t=0 0
m=audio 20262 RTP/AVP 0 8 116 18 101
c=IN IP4 10.118.1.165
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 20678 RTP/AVP 100 126 97
c=IN IP4 10.118.1.165
b=TIAS:4000000
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=640016;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428016;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428016;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=content:main
a=label:11
a=sendrecv
<------------->
--- (22 headers 35 lines) ---
Sending to 10.118.1.165:56009 (no NAT)
Using INVITE request as basis request - 00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165
Found peer '3108' for '3108' from 10.118.1.165:56009
Got SDP version 0 and unique parts [Cisco-SIPUA 14753 IN IP4 10.118.1.165]
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 116
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 116
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Found RTP video format 100
Found RTP video format 126
Found RTP video format 97
Found video description format H264 for ID 100
Found video description format H264 for ID 126
Found video description format H264 for ID 97
Capabilities: us - (ulaw|g729|h264), peer - audio=(ulaw|alaw|g729|ilbc)/video=(h264)/text=(nothing), combined - (ulaw|g729|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.118.1.165:20262
Peer video RTP is at port 10.118.1.165:20678
Looking for 12345678 in from-internal (domain 10.118.1.28)
sip_route_dump: route/path hop: <sip:3108 at 10.118.1.165:56009;transport=tcp>
<--- Transmitting (no NAT) to 10.118.1.165:56009 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.118.1.165:56009;branch=z9hG4bK6087719e;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e76013a71a216aa-2392cbbe
To: <sip:12345678 at 10.118.1.28>
Call-ID: 00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165
CSeq: 102 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer,X-cisco-sis-7.0.0
Contact: <sip:12345678 at 10.118.1.28:5060;transport=tcp>
Call-Info: <urn:x-cisco-remotecc:callinfo>; security=NotAuthenticated
Content-Length: 0
<------------>
[2020-10-25 20:22:44] WARNING[8547][C-00000001]: app_macro.c:274 _macro_exec: Macro() is deprecated and will be removed from a future version of Asterisk.
[2020-10-25 20:22:44] WARNING[8547][C-00000001]: app_macro.c:275 _macro_exec: Dialplan should be updated to use Gosub instead.
[2020-10-25 20:22:44] WARNING[8547][C-00000001]: res_agi.c:2223 launch_script: Failed to execute '/var/www/html/admin/modules/superfecta/agi/superfecta.agi': File does not exist.
[2020-10-25 20:22:44] WARNING[8547][C-00000001]: func_cdr.c:538 cdr_write: Using the CDR function to set 'accountcode' is deprecated. Please use the CHANNEL function instead.
<--- Transmitting (no NAT) to 10.118.1.165:56009 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.118.1.165:56009;branch=z9hG4bK6087719e;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e76013a71a216aa-2392cbbe
To: <sip:12345678 at 10.118.1.28>;tag=as1ed1a366
Call-ID: 00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165
CSeq: 102 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer,X-cisco-sis-7.0.0
Contact: <sip:12345678 at 10.118.1.28:5060;transport=tcp>
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 10.118.1.165:56009 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.118.1.165:56009;branch=z9hG4bK6087719e;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e76013a71a216aa-2392cbbe
To: <sip:12345678 at 10.118.1.28>;tag=as1ed1a366
Call-ID: 00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165
CSeq: 102 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer,X-cisco-sis-7.0.0
Contact: <sip:12345678 at 10.118.1.28:5060;transport=tcp>
Remote-Party-ID: <sip:12345678 at 10.118.1.28>;party=called;privacy=off;screen=no
Call-Info: <urn:x-cisco-remotecc:callinfo>; orientation=to
Content-Length: 0
<------------>
[2020-10-25 20:22:44] ERROR[8216]: res_pjsip_header_funcs.c:410 remove_header: No headers had been previously added to this session.
<--- Transmitting (no NAT) to 10.118.1.165:56009 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.118.1.165:56009;branch=z9hG4bK6087719e;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e76013a71a216aa-2392cbbe
To: <sip:12345678 at 10.118.1.28>;tag=as1ed1a366
Call-ID: 00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165
CSeq: 102 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer,X-cisco-sis-7.0.0
Contact: <sip:12345678 at 10.118.1.28:5060;transport=tcp>
Remote-Party-ID: "CID:" <sip:12345678 at 10.118.1.28>;party=called;privacy=off;screen=no
Call-Info: <urn:x-cisco-remotecc:callinfo>; orientation=to
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 10.118.1.165:56009 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.118.1.165:56009;branch=z9hG4bK6087719e;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e76013a71a216aa-2392cbbe
To: <sip:12345678 at 10.118.1.28>;tag=as1ed1a366
Call-ID: 00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165
CSeq: 102 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer,X-cisco-sis-7.0.0
Contact: <sip:12345678 at 10.118.1.28:5060;transport=tcp>
Call-Info: <urn:x-cisco-remotecc:callinfo>; security=NotAuthenticated
Content-Length: 0
<------------>
Audio is at 18546
Video is at 10.118.1.28:17536
Adding codec ulaw to SDP
Adding codec g729 to SDP
Adding video codec h264 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.118.1.165:56009 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.118.1.165:56009;branch=z9hG4bK6087719e;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e76013a71a216aa-2392cbbe
To: <sip:12345678 at 10.118.1.28>;tag=as1ed1a366
Call-ID: 00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165
CSeq: 102 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer,X-cisco-sis-7.0.0
Contact: <sip:12345678 at 10.118.1.28:5060;transport=tcp>
Remote-Party-ID: "CID:" <sip:12345678 at 10.118.1.28>;party=called;privacy=off;screen=no
Call-Info: <urn:x-cisco-remotecc:callinfo>; orientation=to
Call-Info: <urn:x-cisco-remotecc:callinfo>; security=NotAuthenticated
Content-Type: application/sdp
Content-Length: 581
v=0
o=root 1794875076 1794875076 IN IP4 10.118.1.28
s=Asterisk PBX 16.13.0
c=IN IP4 10.118.1.28
t=0 0
m=audio 18546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 17536 RTP/AVP 100
b=TIAS:4000000
a=rtpmap:100 H264/90000
a=fmtp:100 max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000;packetization-mode=1;level-asymmetry-allowed=1;profile-level-id=640016
a=imageattr:100 recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=sendrecv
<------------>
<--- SIP read from TCP:10.118.1.165:56009 --->
ACK sip:12345678 at 10.118.1.28:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.118.1.165:56009;branch=z9hG4bK0076d261
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e76013a71a216aa-2392cbbe
To: <sip:12345678 at 10.118.1.28>;tag=as1ed1a366
Call-ID: 00cae541-5e760054-4b81d928-3fa56bbd at 10.118.1.165
Max-Forwards: 70
Session-ID: 6c22b06500105000a00000cae5415e76;remote=00000000000000000000000000000000
Date: Mon, 26 Oct 2020 00:22:45 GMT
CSeq: 102 ACK
User-Agent: Cisco-CP-DX650/10.2.5
Content-Length: 0
Recv-Info: conference
Recv-Info: x-cisco-conference
Authorization: Digest username="3108",realm="asterisk",uri="sip:12345678 at 10.118.1.28:5060;transport=tcp",response="e241ab76dd8ee4b13de0148461a6331c",nonce="50842d88",algorithm=MD5
<------------->
--- (14 headers 0 lines) ---
[2020-10-25 20:22:46] WARNING[8547][C-00000001]: chan_sip.c:8455 sip_indicate: Don't know how to indicate condition 36
{noformat}
cisco_usecallmanager=no (both Video & Audio work)
{noformat}
<--- SIP read from TCP:10.118.1.165:39306 --->
SUBSCRIBE sip:12345678 at 10.118.1.28 SIP/2.0
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK5511c3cf
From: <sip:12345678 at 10.118.1.165>;tag=00cae5415e76012507a6ddb9-3b1b0cbd
To: <sip:12345678 at 10.118.1.28>
Call-ID: 00cae541-5e760096-7327422f-5df9bfc0 at 10.118.1.165
Session-ID: 007bf9fd00105000a00000cae5415e76;remote=00000000000000000000000000000000
Date: Mon, 26 Oct 2020 00:20:31 GMT
CSeq: 1000 SUBSCRIBE
User-Agent: Cisco-CP-DX650/10.2.5
Event: presence
Accept: application/cpim-pidf+xml
Expires: 3600
Max-Forwards: 70
Contact: <sip:3108 at 10.118.1.165:39306;transport=tcp>
Authorization: Digest username="3108",realm="asterisk",uri="sip:12345678 at 10.118.1.28",response="e424bc7c0c22014bf58f5c117578286a",nonce="534021a0",algorithm=MD5
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 0
<------------->
--- (17 headers 0 lines) ---
Sending to 10.118.1.165:39306 (NAT)
Creating new subscription
Sending to 10.118.1.165:39306 (NAT)
sip_route_dump: route/path hop: <sip:3108 at 10.118.1.165:39306;transport=tcp>
No matching peer for '12345678' from '10.118.1.165:39306'
Looking for 12345678 in from-sip-external (domain 10.118.1.28)
<--- Transmitting (NAT) to 10.118.1.165:39306 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK5511c3cf;received=10.118.1.165;rport=39306
From: <sip:12345678 at 10.118.1.165>;tag=00cae5415e76012507a6ddb9-3b1b0cbd
To: <sip:12345678 at 10.118.1.28>;tag=as0474f792
Call-ID: 00cae541-5e760096-7327422f-5df9bfc0 at 10.118.1.165
CSeq: 1000 SUBSCRIBE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Content-Length: 0
<------------>
Really destroying SIP dialog '00cae541-5e760096-7327422f-5df9bfc0 at 10.118.1.165' Method: SUBSCRIBE
<--- SIP read from TCP:10.118.1.165:39306 --->
SUBSCRIBE sip:1713443068 at 10.118.1.28 SIP/2.0
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK2b06180d
From: <sip:1713443068 at 10.118.1.165>;tag=00cae5415e760126565732ec-197213b8
To: <sip:1713443068 at 10.118.1.28>
Call-ID: 00cae541-5e760097-62501f04-3ba4822f at 10.118.1.165
Session-ID: 007bf9fd00105000a00000cae5415e76;remote=00000000000000000000000000000000
Date: Mon, 26 Oct 2020 00:20:32 GMT
CSeq: 1000 SUBSCRIBE
User-Agent: Cisco-CP-DX650/10.2.5
Event: presence
Accept: application/cpim-pidf+xml
Expires: 3600
Max-Forwards: 70
Contact: <sip:3108 at 10.118.1.165:39306;transport=tcp>
Authorization: Digest username="3108",realm="asterisk",uri="sip:1713443068 at 10.118.1.28",response="9a47455cafd3d3579913f8f75e49fba2",nonce="534021a0",algorithm=MD5
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 0
<------------->
--- (17 headers 0 lines) ---
Sending to 10.118.1.165:39306 (NAT)
Creating new subscription
Sending to 10.118.1.165:39306 (NAT)
sip_route_dump: route/path hop: <sip:3108 at 10.118.1.165:39306;transport=tcp>
No matching peer for '1713443068' from '10.118.1.165:39306'
Looking for 1713443068 in from-sip-external (domain 10.118.1.28)
<--- Transmitting (NAT) to 10.118.1.165:39306 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK2b06180d;received=10.118.1.165;rport=39306
From: <sip:1713443068 at 10.118.1.165>;tag=00cae5415e760126565732ec-197213b8
To: <sip:1713443068 at 10.118.1.28>;tag=as3f451298
Call-ID: 00cae541-5e760097-62501f04-3ba4822f at 10.118.1.165
CSeq: 1000 SUBSCRIBE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Content-Length: 0
<------------>
<--- SIP read from TCP:10.118.1.165:39306 --->
SUBSCRIBE sip:6580177837 at 10.118.1.28 SIP/2.0
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK66df8d85
From: <sip:6580177837 at 10.118.1.165>;tag=00cae5415e760127564f8d09-3980026e
To: <sip:6580177837 at 10.118.1.28>
Call-ID: 00cae541-5e760098-5e8e9b46-2c539127 at 10.118.1.165
Session-ID: 007bf9fd00105000a00000cae5415e76;remote=00000000000000000000000000000000
Date: Mon, 26 Oct 2020 00:20:32 GMT
CSeq: 1000 SUBSCRIBE
User-Agent: Cisco-CP-DX650/10.2.5
Event: presence
Accept: application/cpim-pidf+xml
Expires: 3600
Max-Forwards: 70
Contact: <sip:3108 at 10.118.1.165:39306;transport=tcp>
Authorization: Digest username="3108",realm="asterisk",uri="sip:6580177837 at 10.118.1.28",response="7fce5d9e09f5d17f062c0c6ce28a1759",nonce="534021a0",algorithm=MD5
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 0
<------------->
--- (17 headers 0 lines) ---
Sending to 10.118.1.165:39306 (NAT)
Creating new subscription
Sending to 10.118.1.165:39306 (NAT)
sip_route_dump: route/path hop: <sip:3108 at 10.118.1.165:39306;transport=tcp>
No matching peer for '6580177837' from '10.118.1.165:39306'
Looking for 6580177837 in from-sip-external (domain 10.118.1.28)
<--- Transmitting (NAT) to 10.118.1.165:39306 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK66df8d85;received=10.118.1.165;rport=39306
From: <sip:6580177837 at 10.118.1.165>;tag=00cae5415e760127564f8d09-3980026e
To: <sip:6580177837 at 10.118.1.28>;tag=as55aa2954
Call-ID: 00cae541-5e760098-5e8e9b46-2c539127 at 10.118.1.165
CSeq: 1000 SUBSCRIBE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Content-Length: 0
<------------>
Really destroying SIP dialog '00cae541-5e760097-62501f04-3ba4822f at 10.118.1.165' Method: SUBSCRIBE
Really destroying SIP dialog '00cae541-5e760098-5e8e9b46-2c539127 at 10.118.1.165' Method: SUBSCRIBE
<--- SIP read from TCP:10.118.1.165:39306 --->
INVITE sip:12345678 at 10.118.1.28;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK20fcbe81
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
Max-Forwards: 70
Session-ID: 1f5a264800105000a00000cae5415e76;remote=00000000000000000000000000000000
Date: Mon, 26 Oct 2020 00:20:34 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP-DX650/10.2.5
Contact: <sip:3108 at 10.118.1.165:39306;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP00CAE5415E76";video
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Authorization: Digest username="3108",realm="asterisk",uri="sip:12345678 at 10.118.1.28;user=phone",response="5dc8f4698f80152cdad101b18981c1c2",nonce="534021a0",algorithm=MD5
Content-Length: 757
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 2398 0 IN IP4 10.118.1.165
s=SIP Call
b=AS:4064
t=0 0
m=audio 17466 RTP/AVP 0 8 116 18 101
c=IN IP4 10.118.1.165
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 17800 RTP/AVP 97
c=IN IP4 10.118.1.165
b=TIAS:4000000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428016;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=content:main
a=label:11
a=sendrecv
<------------->
--- (22 headers 29 lines) ---
Sending to 10.118.1.165:39306 (NAT)
Sending to 10.118.1.165:39306 (NAT)
Using INVITE request as basis request - 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
Found peer '3108' for '3108' from 10.118.1.165:39306
<--- Reliably Transmitting (no NAT) to 10.118.1.165:39306 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK20fcbe81;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>;tag=as33691968
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
CSeq: 101 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2a1fe1b9"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165' in 6400 ms (Method: INVITE)
<--- SIP read from TCP:10.118.1.165:39306 --->
ACK sip:12345678 at 10.118.1.28;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK20fcbe81
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>;tag=as33691968
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
Session-ID: 1f5a264800105000a00000cae5415e76;remote=00000000000000000000000000000000
Max-Forwards: 70
Date: Mon, 26 Oct 2020 00:20:34 GMT
CSeq: 101 ACK
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from TCP:10.118.1.165:39306 --->
INVITE sip:12345678 at 10.118.1.28;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK70ea52f8
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
Max-Forwards: 70
Session-ID: 1f5a264800105000a00000cae5415e76;remote=00000000000000000000000000000000
Date: Mon, 26 Oct 2020 00:20:34 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP-DX650/10.2.5
Contact: <sip:3108 at 10.118.1.165:39306;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP00CAE5415E76";video
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Authorization: Digest username="3108",realm="asterisk",uri="sip:12345678 at 10.118.1.28;user=phone",response="96e2522a409528a76e970fe9c261ecd3",nonce="2a1fe1b9",algorithm=MD5
Content-Length: 757
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 2398 0 IN IP4 10.118.1.165
s=SIP Call
b=AS:4064
t=0 0
m=audio 17466 RTP/AVP 0 8 116 18 101
c=IN IP4 10.118.1.165
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 17800 RTP/AVP 97
c=IN IP4 10.118.1.165
b=TIAS:4000000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428016;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=content:main
a=label:11
a=sendrecv
<------------->
--- (22 headers 29 lines) ---
Sending to 10.118.1.165:39306 (no NAT)
Using INVITE request as basis request - 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
Found peer '3108' for '3108' from 10.118.1.165:39306
Got SDP version 0 and unique parts [Cisco-SIPUA 2398 IN IP4 10.118.1.165]
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 116
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 116
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Found RTP video format 97
Found video description format H264 for ID 97
Capabilities: us - (ulaw|g729|h264), peer - audio=(ulaw|alaw|g729|ilbc)/video=(h264)/text=(nothing), combined - (ulaw|g729|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.118.1.165:17466
Peer video RTP is at port 10.118.1.165:17800
Looking for 12345678 in from-internal (domain 10.118.1.28)
sip_route_dump: route/path hop: <sip:3108 at 10.118.1.165:39306;transport=tcp>
<--- Transmitting (no NAT) to 10.118.1.165:39306 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK70ea52f8;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
CSeq: 102 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:12345678 at 10.118.1.28:5060;transport=tcp>
Content-Length: 0
<------------>
[2020-10-25 20:20:35] WARNING[7888][C-00000001]: app_macro.c:274 _macro_exec: Macro() is deprecated and will be removed from a future version of Asterisk.
[2020-10-25 20:20:35] WARNING[7888][C-00000001]: app_macro.c:275 _macro_exec: Dialplan should be updated to use Gosub instead.
[2020-10-25 20:20:35] WARNING[7888][C-00000001]: res_agi.c:2223 launch_script: Failed to execute '/var/www/html/admin/modules/superfecta/agi/superfecta.agi': File does not exist.
[2020-10-25 20:20:35] WARNING[7888][C-00000001]: func_cdr.c:538 cdr_write: Using the CDR function to set 'accountcode' is deprecated. Please use the CHANNEL function instead.
<--- Transmitting (no NAT) to 10.118.1.165:39306 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK70ea52f8;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>;tag=as6f0b29db
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
CSeq: 102 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:12345678 at 10.118.1.28:5060;transport=tcp>
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 10.118.1.165:39306 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK70ea52f8;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>;tag=as6f0b29db
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
CSeq: 102 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:12345678 at 10.118.1.28:5060;transport=tcp>
Remote-Party-ID: <sip:12345678 at 10.118.1.28>;party=called;privacy=off;screen=no
Content-Length: 0
<------------>
[2020-10-25 20:20:35] ERROR[7546]: res_pjsip_header_funcs.c:410 remove_header: No headers had been previously added to this session.
<--- Transmitting (no NAT) to 10.118.1.165:39306 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK70ea52f8;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>;tag=as6f0b29db
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
CSeq: 102 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:12345678 at 10.118.1.28:5060;transport=tcp>
Remote-Party-ID: "CID:" <sip:12345678 at 10.118.1.28>;party=called;privacy=off;screen=no
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 10.118.1.165:39306 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK70ea52f8;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>;tag=as6f0b29db
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
CSeq: 102 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:12345678 at 10.118.1.28:5060;transport=tcp>
Content-Length: 0
<------------>
Audio is at 16442
Video is at 10.118.1.28:19258
Adding codec ulaw to SDP
Adding codec g729 to SDP
Adding video codec h264 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.118.1.165:39306 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK70ea52f8;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>;tag=as6f0b29db
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
CSeq: 102 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:12345678 at 10.118.1.28:5060;transport=tcp>
Remote-Party-ID: "CID:" <sip:12345678 at 10.118.1.28>;party=called;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 509
v=0
o=root 257785350 257785350 IN IP4 10.118.1.28
s=Asterisk PBX 16.13.0
c=IN IP4 10.118.1.28
b=CT:40000000
t=0 0
m=audio 16442 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 19258 RTP/AVP 97
a=rtpmap:97 H264/90000
a=fmtp:97 max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000;packetization-mode=0;level-asymmetry-allowed=1;profile-level-id=428016
a=sendrecv
<------------>
<--- SIP read from TCP:10.118.1.165:39306 --->
ACK sip:12345678 at 10.118.1.28:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK72aec41c
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>;tag=as6f0b29db
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
Max-Forwards: 70
Session-ID: 1f5a264800105000a00000cae5415e76;remote=00000000000000000000000000000000
Date: Mon, 26 Oct 2020 00:20:36 GMT
CSeq: 102 ACK
User-Agent: Cisco-CP-DX650/10.2.5
Content-Length: 0
Recv-Info: conference
Recv-Info: x-cisco-conference
Authorization: Digest username="3108",realm="asterisk",uri="sip:12345678 at 10.118.1.28:5060;transport=tcp",response="a57be16986998cda7534cc6bfb002a62",nonce="2a1fe1b9",algorithm=MD5
<------------->
--- (14 headers 0 lines) ---
<--- SIP read from TCP:10.118.1.165:39306 --->
INFO sip:12345678 at 10.118.1.28:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK66865a42
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>;tag=as6f0b29db
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
Max-Forwards: 70
Session-ID: 1f5a264800105000a00000cae5415e76;remote=00000000000000000000000000000000
Date: Mon, 26 Oct 2020 00:20:36 GMT
CSeq: 103 INFO
User-Agent: Cisco-CP-DX650/10.2.5
Contact: <sip:3108 at 10.118.1.165:39306;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP00CAE5415E76";video
Authorization: Digest username="3108",realm="asterisk",uri="sip:12345678 at 10.118.1.28:5060;transport=tcp",response="a9fd528f3237a2f49bb45194c1d16591",nonce="2a1fe1b9",algorithm=MD5
Content-Length: 243
Content-Type: application/media_control+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8" standalone="no" ?>
<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update/>
</to_encoder>
</vc_primitive>
</media_control>
<------------->
--- (15 headers 8 lines) ---
Receiving INFO!
<--- Transmitting (no NAT) to 10.118.1.165:39306 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK66865a42;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>;tag=as6f0b29db
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
CSeq: 103 INFO
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Content-Length: 0
<------------>
<--- SIP read from TCP:10.118.1.165:39306 --->
INFO sip:12345678 at 10.118.1.28:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK5d7b11ac
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>;tag=as6f0b29db
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
Max-Forwards: 70
Session-ID: 1f5a264800105000a00000cae5415e76;remote=00000000000000000000000000000000
Date: Mon, 26 Oct 2020 00:20:41 GMT
CSeq: 104 INFO
User-Agent: Cisco-CP-DX650/10.2.5
Contact: <sip:3108 at 10.118.1.165:39306;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP00CAE5415E76";video
Authorization: Digest username="3108",realm="asterisk",uri="sip:12345678 at 10.118.1.28:5060;transport=tcp",response="a9fd528f3237a2f49bb45194c1d16591",nonce="2a1fe1b9",algorithm=MD5
Content-Length: 243
Content-Type: application/media_control+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8" standalone="no" ?>
<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update/>
</to_encoder>
</vc_primitive>
</media_control>
<------------->
--- (15 headers 8 lines) ---
Receiving INFO!
<--- Transmitting (no NAT) to 10.118.1.165:39306 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.118.1.165:39306;branch=z9hG4bK5d7b11ac;received=10.118.1.165
From: "3108" <sip:3108 at 10.118.1.28>;tag=00cae5415e760128324a48cc-595c80e7
To: <sip:12345678 at 10.118.1.28>;tag=as6f0b29db
Call-ID: 00cae541-5e760052-796cf9da-3deca4a3 at 10.118.1.165
CSeq: 104 INFO
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Content-Length: 0
<------------>
{noformat}
> [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML
> -------------------------------------------------------------------------------
>
> Key: ASTERISK-13145
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-13145
> Project: Asterisk
> Issue Type: New Feature
> Components: Channels/chan_sip/NewFeature
> Reporter: Gareth Palmer
> Assignee: Gareth Palmer
> Labels: patch, pjsip
> Attachments: 00_READ_ME_FIRST.txt, AppDialRules.xml, cisco-usecallmanager-13.36.0.patch, cisco-usecallmanager-16.13.0.patch, DialTemplate.xml, FeaturePolicy.xml, SEPMAC.cnf.xml, SoftKeys.xml, usecallmanager-port.sh
>
>
> This patch provides support for Cisco 6900, 7900, 8800 and 9900 series phones using the SIP firmware.
> Available features are: Busy Lamp Field, Off Hook Notification, Call Forward, Do Not Disturb, Huntgroup Login, Call Park (Notify and Monitor), Server-Side Ad-Hoc Conference, Conference List, Kick and Mute/Unmute, Multi-Admin Conference, Multiple Lines via Bulk Register, Immediate Divert, Call Recording, Restart or Reset via CLI, Call Pickup Notification, Call Back, Join Calls, Mallicious Call ID, Quality Reporting Tool and Fail-over/Fail-back.
> Also included is Application Server Events used by non-USECALLMANAGER phones (Call Forward and Do Not Disturb only).
> *Important:* Read the documentation at [http://usecallmanager.nz] to see the additional configuration options required for the phones to operate correctly.
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